Originally posted by Larian
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Enter Veromix and PulseAudio. After installing Yate and during the first phone call, I can use controls in Veromix to change Yate's audio routing. I can send output to and take input from the headset. Fortunately, I only need to do this once after installing, as the configuration is saved. Furthermore, if I want to switch to speakerphone-like behavior, I can route the output and input back to the laptop's hardware.
Agreed, this is suboptimal. Better would be for Yate to incorporate some audio configuration, like just about every other softphone I've tried. But, alas, I can't get any of those to properly perform (or hold) a SIP registration. The problem lies somewhere, I think, in Avaya's SIP implementation. But I'm no SIP expert, so I'm kind of guessing here.
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